search for: spa942

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short t...

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I m...

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample...

Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw regarding this feature...

currently using 1.2.14 and zaptel 1.2.12 i'm using mfc/r2 so i can't move to 1.4 with sip jitter control and improved jitter control in zaptel 1.4. my problem is excessive jitter using linksys spa942. when i set canreinvite=no, which forces rtp to pass through *, quality is horrible. clicking sounds, pauses, etc. but when omitted or canreinvite=yes, sip to sip calls are ok. now, the problem comes to zap calls, i have a te110p using unicall mfc/r2, since rtp passes through *, quality is agai...

I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set that in my sip.conf file as well: context...

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring a...

Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11 21:37:31] -- Cal...

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

...xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca29d at 10.1.24.145^M CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: "sss-sss-ssss" ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PC...

...base station or the handset - to extensions within my office as well as numbers outside the network. But I can't receive calls on either the base station or the handset. All of the calls go strait to voice mail. I've never had this problem with the phones I use in my office - Linksys SPA942. What am I doing wrong? Thanks, Danny

...utton * Press "xferLx" to join those to calls. This works everywhere except on the one behind NAT. After a lot of messing around with all the options possible I gave up and subscribed here... As of now, the only thing I've found out is that when you press the "xferLx" the SPA942 sends a "REFER" message to Asterisk, but there's one thing that seems wrong, the "Via" header. On every message that reaches Asterisk the Via header is: incoming: Via: SIP/2.0/UDP <WAN IP>:18363;branch=z9hG4bK-bb6024d6 outgoing: Via: SIP/2.0/UDP <WAN IP>:18363...

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring a...

I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080326/2fd2c557/attachment.htm

Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches "fix" phones, the correspondant can't hear the caller. What is very stange is that for incom...

...a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but probably good enough). Can I set one of these or a similar Cisco phone to auto answer in speaker mode? Any ideas on an alternative phone that would allow this? The alternative is to just set up the call locally and then leave the room with t...

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik